similar to: End User guide

Actually, I have tested this here with an Aastra 9133i and an Asterisk@Home server, and the 9133i will re-subscribe on its own after an Asterisk reboot, if you wait long enough. It took on the order of an hour to do so. Of course, a phone reboot will get it done faster, if necessary, but it _will_ eventually re-subscribe on its own. > Thanks Olle, > > So am I to understand that you

I'd recommend any of the following, which are all in your price range Snom 300 Polycom IP430 Polycom IP501 Aastra 9112i Linksys SPA-922 Grandstream GXP-2000 Cory Andrews ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Wednesday, November 01, 2006 11:17 AM To: Asterisk

After working with the Grandstream GXP 2000 series phones, I have decided that I am quite unhappy with their problems, both voice quality, volume, features and others. For their price now, there are plenty of phones to choose from as well. So subjectively what would be the best Hardphone for a small/medium business with multiple line support, BLF, etc. Are the Cisco 7960 the best of the

There is no release note, just a text file that says AASTRA TELECOM INC. February 2006 FC-000046-01-07.st - 9133i Generic SIP Firmware 1.3.1.1095 for customer release. ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gareth Owen Sent: 15 February 2006 02:00 To: asterisk-users@lists.digium.com

I am unable to get my 9133i to register with my asterisk server. I am including config files below, this a simple test network so there's nothing secret in the config files. I have upgraded the phone to the latest software version (1.4.3) I'm not sure what the problem is. I can call the phone from a softphone, but the 9133i says "no service" on the screen and I can't dial

Hi there, I am running Asterisk 1.2. I have a Grandstream GXP2000 and Aastra 9133i with BLF/Speedial configured for other extensions. The hint's are all configured in extensions.conf and it seems to work as it is supposed to until I reload the configuration in Asterisk or reboot the server. Then neither phone displays the BLF of other extensions until I reboot the phone. It continues

I have upgraded the GXP-2000 to the newest firmware 1.0.1.12<http://1.0.1.12>and the phone is much more usable However, I still have two slight sound quality issues: 1) There is static on the line at all times. It is not that noticable to me, but when I make calls out the PSTN the person on the other end hears it. If I use a Cisco ATA with an analog phone and call the same person again

I've been pulling out my hair all day on this one. If anyone can help, I'd really appreciate it. :-( I've got an Aastra 9133i (with the latest firmware version) and a Cisco 7960 sitting behind a NAT device on my LAN. The Asterisk server is hosted offsite and has a public IP address. I've set up port-forwarding on the firewall for both phones to tunnel the SIP messages initiated

Hi When I make a call with the Grandstream GXP-2000 through Asterisk (and SER) to landline using VSP, after I hang up the call the other party are still connected for another 30-40 seconds. I've notice that the SIP BYE is sent to Asterisk, but Asterisk sends no SIP BYE on to VSP. When I use the SPA-941 the call terminates on the other right away soon as I hang up. I have updated the

FYI, spoke with Grandstream this morning, the GXP-2000 release has been delayed again. Looking like April now before these hit the street. -- Cory Andrews Senior Partner VOIPSupply.com +++++++++++++ V: 800.398.VOIP X22 F: 716.630.1548 E: Cory@VOIPSupply.com

We have several types of phones, Cisco 79xx, Aastra 9133i etc. We have a "hotdesk" type system where anyone can log on to an extension - however what I would love to do is relabel the phone with the current "owner" when this logon happens. I know that I can change the sip.conf and phones tftp file, however this is a big problem with the Cisco's as they take *forever* (ok,

Is there any such thing is a userguide for asterisk from an enduser point of view ie. what to do to transfer a call etc ? I've looked through all the official documentation and nothing exists, and trying to install an ASterisk at a client can't even explain how to transfer a call to another extension! Even some basic help would be welcome! cheers Dave

Hello - I have just purchased an Aastra 9133i SIP phone for testing with Asterisk. Its a little flakey but overall is a far superior phone to the others in the $179 range. I have an issue regarding the message waiting indicator. The phone does not seem to respond to the "NOTIFY" command from Asterisk. Searching archives seems to indicate that this was previously an issue on the 480i

Hello - I have just purchased an Aastra 9133i SIP phone for testing with Asterisk. Its a little flakey but overall is a far superior phone to the others in the $179 range. I have an issue regarding the message waiting indicator. The phone does not seem to respond to the "NOTIFY" command from Asterisk. Searching archives seems to indicate that this was previously an issue on the

Ken - take a look at using IAX protocol to route calls between your Asterisk boxes. Cory Andrews -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ken Williams Sent: Wednesday, November 01, 2006 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [SPAM HEADER] - RE: [asterisk-users]

Thanks everyone for the input. After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over. I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no home phone (just cell phones) so this'll be a great way of testing. All

Thank you Melissa. I love the phone but the dial keypad is a little bouncy. I was hoping for a more solid feel like on the analog PT390's or my quality standard, the Nortel 9417CW. Other than the MWI problem, I'd like more documentation on the configuration paramters. I have found little online configuration documentation other than very basic stuff on the Sayson website. I'd

I upgraded my office server from 1.4.22 to 1.6.0.5 on the weekend and after some testing there seems to be a compatibility problem when using Aastra phones. With 1.6.0.5 all incoming calls to all Aastra phones were dropped after a minute or so. I installed 1.6.1-rc1 and now the newer Aastra phones (5xi) work properly. The problem remains with the older phones (9112i, 9133i and 480i). If I dial

Actually, I have tested this here with an Aastra 9133i and an Asterisk@Home server, and the 9133i will re-subscribe on its own after an Asterisk reboot, if you wait long enough. It took on the order of an hour to do so. Of course, a phone reboot will get it done faster, if necessary, but it _will_ eventually re-subscribe on its own. In another thread, I've seen a response that the GXP2000

Aastra Telecom has released SIP v1.3.1 firmware for the Aastra range of IP phones (480i, 480iCT, 9112i and 9133i). The firmware and release notes (no updated admin and user guides yet) are available for download at: http://www.aastra.com/support/enterpriseip Contrary to what the version numbering would suggest, this is a significant update with many new features and bug fixes. See the release